From bcf29dc6707e55f7b82f244f05ef5fbd2e008f6b Mon Sep 17 00:00:00 2001
From: Mark Furneaux <mark@furneaux.ca>
Date: Sun, 15 May 2016 13:31:00 -0400
Subject: [PATCH] Fix caps on audio pipeline

Fixes #1747

The pipeline has the caps for the analyzer applied in the wrong place. This results in the audio output being limited to 16 bit regardless of the input file.
This change also cleans up the mono/sample rate caps as well.
---
 src/engines/gstenginepipeline.cpp | 45 ++++++++++++++-----------------
 1 file changed, 20 insertions(+), 25 deletions(-)

diff --git a/src/engines/gstenginepipeline.cpp b/src/engines/gstenginepipeline.cpp
index 2340fc513..a6e270f4b 100644
--- a/src/engines/gstenginepipeline.cpp
+++ b/src/engines/gstenginepipeline.cpp
@@ -393,13 +393,7 @@ bool GstEnginePipeline::Init() {
   }
 
   gst_element_link_many(queue_, audioconvert_, convert_sink, nullptr);
-
-  // Link the elements with special caps
-  // The scope path through the tee gets 16-bit ints.
-  GstCaps* caps16 = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING,
-                                        "S16LE", NULL);
-  gst_element_link_filtered(probe_converter, probe_sink, caps16);
-  gst_caps_unref(caps16);
+  gst_element_link(probe_converter, probe_sink);
 
   // Link the outputs of tee to the queues on each path.
   gst_pad_link(gst_element_get_request_pad(tee, "src_%u"),
@@ -412,31 +406,32 @@ bool GstEnginePipeline::Init() {
     gst_element_link_many(rgvolume_, rglimiter_, audioconvert2_, tee, nullptr);
   }
 
-  // Link everything else.
-  gst_element_link(probe_queue, probe_converter);
+  // Link the analyzer output of the tee and force 16 bit caps
+  GstCaps* caps16 = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING,
+                                        "S16LE", NULL);
+  gst_element_link_filtered(probe_queue, probe_converter, caps16);
+  gst_caps_unref(caps16);
+
   gst_element_link_many(audio_queue, equalizer_preamp_, equalizer_,
                         stereo_panorama_, volume_, audioscale_, convert,
                         nullptr);
 
-  // add caps for fixed sample rate and mono, but only if requested
-  if (sample_rate_ != GstEngine::kAutoSampleRate && sample_rate_ > 0) {
-    GstCaps* caps = gst_caps_new_simple("audio/x-raw", "rate", G_TYPE_INT,
-                                        sample_rate_, nullptr);
-    if (mono_playback_) {
-      gst_caps_set_simple(caps, "channels", G_TYPE_INT, 1, nullptr);
-    }
+  // Ensure that the audio output of the tee does not autonegotiate to 16 bit
+  GstCaps* caps = gst_caps_new_simple("audio/x-raw", "format", G_TYPE_STRING,
+                                      "F32LE", NULL);
 
-    gst_element_link_filtered(convert, audiosink_, caps);
-    gst_caps_unref(caps);
-  } else if (mono_playback_) {
-    GstCaps* capsmono =
-        gst_caps_new_simple("audio/x-raw", "channels", G_TYPE_INT, 1, nullptr);
-    gst_element_link_filtered(convert, audiosink_, capsmono);
-    gst_caps_unref(capsmono);
-  } else {
-    gst_element_link(convert, audiosink_);
+  // Add caps for fixed sample rate and mono, but only if requested
+  if (sample_rate_ != GstEngine::kAutoSampleRate && sample_rate_ > 0) {
+    gst_caps_set_simple(caps, "rate", G_TYPE_INT, sample_rate_, nullptr);
   }
 
+  if (mono_playback_) {
+    gst_caps_set_simple(caps, "channels", G_TYPE_INT, 1, nullptr);
+  }
+
+  gst_element_link_filtered(convert, audiosink_, caps);
+  gst_caps_unref(caps);
+
   // Add probes and handlers.
   gst_pad_add_probe(gst_element_get_static_pad(probe_converter, "src"),
                     GST_PAD_PROBE_TYPE_BUFFER, HandoffCallback, this, nullptr);